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...M, Thyda ENG <engthyda at gmail.com> wrote: > According to what I have done , I add the message_context to the > pjsip.endpoint_custom.conf in /etc/asterisk and then I create that > message_context in the extension.conf, and it works. > > On Tue, Nov 17, 2015 at 9:34 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> So, the only thing that is needed in the endpoint definition in >> pjsip.conf (there is no such file pjsip.endpoint_custom.conf) is >> >> *message_context=astsms* >> >> Is that correct? Anything I need t...

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am able to make > "from-interna...

...t; > use_avpf=no > > ice_support=no > > media_use_received_transport=no > > trust_id_inbound=yes > > media_encryption=no > > rtp_symmetric=yes > > rewrite_contact=yes > > *message_context=astsms* > > > > On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Hello, >> >> I am looking for documentation support for enabling instant messaging >> between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as >> Zoiper. Where do I enable this support on the s...

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > > OK. Let me ask this. Is anything else necessary, except choosing TCP as > the > > preferred protocol on the client, to make TCP w Asterisk work? At the > > moment, I have only changed one line in pjsip.conf from my working UDP > > setup: > > > > [tr...

...-555-1212 does not ring. at hangup on caller (sonny): == Spawn extension (from-internal, 912025551212, 2) exited non-zero on 'PJSIP/sonny-00000031' On Sun, Mar 15, 2015 at 3:25 PM, George Joseph <george.joseph at fairview5.com> wrote: > On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> That was the issue, thanks. I now am able to get the caller ringing on an >> outbound call, but an external phone number (E164) I am dialing does not >> ring. >> > > Any error messages? If you set 'core s...

...was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic >> configuration works, and I am connected to a SIP trunk using SIP.US, and >> have set up my inbound calling which works correctly (when I call my PB...

...e to > send a call to an extension where it is behind NAT, Asterisk must update > the contact to have the current IP and port that the phone registered via > (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state > for). > > On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I am following the instructions in >> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I >> am trying to make a call from extension Alice (6001) to extension for Bob >> (6002). When I make the...

...osing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ; <--------------- only this line was changed. On Wed, Feb 17, 2016 at 8:28 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > OK. I will report with my findings. It appears increasingly likely that I > have done something very silly on my side. It is a little perplexing that > the EXACT setup (on the same machine) worked for UDP ... > > On Wed, Feb 17, 2016...

Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and "see" them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Yes, I think the dial does get executed (sonny calling outbound > 202-555-1212): > > core set verbose 3 > Console verbose was OFF and is now 3. > -- Executing [912025551212 at from-internal:1] > Log("PJSIP/sonny-0000003...

Nope, there are no contacts to show that pertain to these endpoints (only my SIP trunks show up). On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > >> Does this help: >> > > Yes, the transport parameter is in the Contact header so it's interesting > it didn't work. If you use pjsip show contacts what is the contact for the > AOR? > > > -- > Joshua Colp > Digium, Inc. | Senior Software...

...calls into my context (fromgw). Unfortunately, the actual caller ID (6175551212) is not getting passed (but I know Asterisk is getting this). How do I "reap" this actual caller ID in my dialplan? On Fri, Mar 13, 2015 at 4:55 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > [sonnyGW1] >> type=identity >> endpoint=sonnyGW1 >> match=65.254.44.194 >> > > You want type=identify, not type=identity. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan...

...long ago, I removed pjsip.conf-based Twilio config and placed it all in pjsip_wizard.conf. Thanks, re: wiki, I will be using it heavily, for sure ;-) On Thu, Feb 18, 2016 at 9:56 PM, George Joseph <george.joseph at fairview5.com> wrote: > > > On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Hello, >> >> I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio >> gateway. I am able to make calls outbound through the gateway, but I am not >> able to make calls into the PBX from external PS...

Sonny Rajagopalan wrote: > Sorry, I was not being very clear, Joshua, and thanks for your patience > with this issue. > > I had set pjsip set logger on and core set debug 99. See absolutely > zilch on asterisk CLI. Or in /var/log/asterisk/messages. If the messages > are not reaching Asterisk, what...

I tried to create a domain with drbd based disk as the backend "drbd:<resourcename>,..," and I get a "unknown disk type" error. I can spin out a patch if needed, but just wanted to know apriori if this was left out for any specific reason. shriram _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com

This patch series adds support for network buffering in the Remus codebase in libxl. Changes in V3: [1/5] Fix redundant checks in configure scripts (based on Ian Campbell''s suggestions) [2/5] Introduce locking in the script, during IFB setup. Add xenstore paths used by netbuf scripts to xenstore-paths.markdown [3/5] Hotplug scripts setup/teardown invocations are now

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say > > transport=tcp ; the only...

On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > That was the issue, thanks. I now am able to get the caller ringing on an > outbound call, but an external phone number (E164) I am dialing does not > ring. > Any error messages? If you set 'core set verbose 3' and try it, does...

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf